# webrtcbin

This webrtcbin implements the majority of the W3's peerconnection API and implementation guide where possible. Generating offers, answers and setting local and remote SDP's are all supported. Both media descriptions and descriptions involving data channels are supported.

Each input/output pad is equivalent to a Track in W3 parlance which are added/removed from the bin. The number of requested sink pads is the number of streams that will be sent to the receiver and will be associated with a GstWebRTCRTPTransceiver (very similar to W3 RTPTransceiver's).

On the receiving side, RTPTransceiver's are created in response to setting a remote description. Output pads for the receiving streams in the set description are also created when data is received.

A TransportStream is created when needed in order to transport the data over the necessary DTLS/ICE channel to the peer. The exact configuration depends on the negotiated SDP's between the peers based on the bundle and rtcp configuration. Some cases are outlined below for a simple single audio/video/data session:

- max-bundle uses a single transport for all media/data transported. Renegotiation involves adding/removing the necessary streams to the existing transports.
- max-compat involves two TransportStream per media stream to transport the rtp and the rtcp packets and a single TransportStream for all data channels. Each stream change involves modifying the associated TransportStream/s as necessary.

## Hierarchy

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstElement
                    ╰──GstBin
                        ╰──webrtcbin

## Implemented interfaces

    GstChildProxy

## Factory details

Authors: – Matthew Waters <matthew></matthew>

Classification: – `Filter/Network/WebRTC`

Rank – primary

Plugin – webrtc

Package – GStreamer Bad Plug-ins

## Pad Templates

### `sink_%u`

    application/x-rtp:

Presence – request

Direction – sink

Object type – GstWebRTCBinSinkPad

### `src_%u`

    application/x-rtp:

Presence – sometimes

Direction – src

Object type – GstWebRTCBinSrcPad

## Signals

### on-data-channel

    on_data_channel_callback (GstElement * object,
                              GstWebRTCDataChannel * channel,
                              gpointer udata)

    def on_data_channel_callback (object, channel, udata):
        #python callback for the 'on-data-channel' signal

    function on_data_channel_callback(object: GstElement * object, channel: GstWebRTCDataChannel * channel, udata: gpointer udata): {
        // javascript callback for the 'on-data-channel' signal
    }

Parameters:

`object` –

the webrtcbin

`channel` –

the new `GstWebRTCDataChannel`

`udata` –
No description available

Flags: Run Last

### on-ice-candidate

    on_ice_candidate_callback (GstElement * object,
                               guint mline_index,
                               gchararray candidate,
                               gpointer udata)

    def on_ice_candidate_callback (object, mline_index, candidate, udata):
        #python callback for the 'on-ice-candidate' signal

    function on_ice_candidate_callback(object: GstElement * object, mline_index: guint mline_index, candidate: gchararray candidate, udata: gpointer udata): {
        // javascript callback for the 'on-ice-candidate' signal
    }

Parameters:

`object` –

the webrtcbin

`mline_index` –

the index of the media description in the SDP

`candidate` –

the ICE candidate

`udata` –
No description available

Flags: Run Last

### on-negotiation-needed

    on_negotiation_needed_callback (GstElement * object,
                                    gpointer udata)

    def on_negotiation_needed_callback (object, udata):
        #python callback for the 'on-negotiation-needed' signal

    function on_negotiation_needed_callback(object: GstElement * object, udata: gpointer udata): {
        // javascript callback for the 'on-negotiation-needed' signal
    }

Parameters:

`object` –

the webrtcbin

`udata` –
No description available

Flags: Run Last

### on-new-transceiver

    on_new_transceiver_callback (GstElement * object,
                                 GstWebRTCRTPTransceiver * candidate,
                                 gpointer udata)

    def on_new_transceiver_callback (object, candidate, udata):
        #python callback for the 'on-new-transceiver' signal

    function on_new_transceiver_callback(object: GstElement * object, candidate: GstWebRTCRTPTransceiver * candidate, udata: gpointer udata): {
        // javascript callback for the 'on-new-transceiver' signal
    }

Parameters:

`object` –

the webrtcbin

`candidate` –

the new GstWebRTCRTPTransceiver

`udata` –
No description available

Flags: Run Last

### prepare-data-channel

    prepare_data_channel_callback (GstElement * object,
                                   GstWebRTCDataChannel * channel,
                                   gboolean is_local,
                                   gpointer udata)

    def prepare_data_channel_callback (object, channel, is_local, udata):
        #python callback for the 'prepare-data-channel' signal

    function prepare_data_channel_callback(object: GstElement * object, channel: GstWebRTCDataChannel * channel, is_local: gboolean is_local, udata: gpointer udata): {
        // javascript callback for the 'prepare-data-channel' signal
    }

Allows data-channel consumers to configure signal handlers on a newly created data-channel, before any data or state change has been notified.

Parameters:

`object` –

the webrtcbin

`channel` –

the new `GstWebRTCDataChannel`

`is_local` –

Whether this channel is local or remote

`udata` –
No description available

Flags: Run Last

Since : 1.22

### request-aux-sender

    GstElement *
    request_aux_sender_callback (GstElement * object,
                                 GstWebRTCDTLSTransport * dtls-transport,
                                 gpointer udata)

    def request_aux_sender_callback (object, dtls-transport, udata):
        #python callback for the 'request-aux-sender' signal

    function request_aux_sender_callback(object: GstElement * object, dtls-transport: GstWebRTCDTLSTransport * dtls-transport, udata: gpointer udata): {
        // javascript callback for the 'request-aux-sender' signal
    }

Request an AUX sender element for the given dtls-transport.

Parameters:

`object` –

the webrtcbin

`dtls-transport` –

The GstWebRTCDTLSTransport object for which the aux sender will be used.

`udata` –
No description available
Returns (GstElement *) –

A new GStreamer element

Flags: Run Last

Since : 1.22

### request-post-rtp-aux-sender

    GstElement *
    request_post_rtp_aux_sender_callback (GstElement * object,
                                          GstWebRTCDTLSTransport * dtls-transport,
                                          gpointer udata)

    def request_post_rtp_aux_sender_callback (object, dtls-transport, udata):
        #python callback for the 'request-post-rtp-aux-sender' signal

    function request_post_rtp_aux_sender_callback(object: GstElement * object, dtls-transport: GstWebRTCDTLSTransport * dtls-transport, udata: gpointer udata): {
        // javascript callback for the 'request-post-rtp-aux-sender' signal
    }

Request an AUX sender element for the given dtls-transport, that is placed between RTPBin and Transportbin..

Parameters:

`object` –

the webrtcbin

`dtls-transport` –

The GstWebRTCDTLSTransport object for which the aux sender will be used.

`udata` –
No description available
Returns (GstElement *) –

A new GStreamer element

Flags: Run Last

Since : 1.26

## Action Signals

### add-ice-candidate

    g_signal_emit_by_name (object, "add-ice-candidate", mline_index, ice-candidate);

    ret = object.emit ("add-ice-candidate", mline_index, ice-candidate)

    let ret = object.emit ("add-ice-candidate", mline_index, ice-candidate);

Parameters:

`object` (GstElement *) –

the webrtcbin

`mline_index` (guint) –

the index of the media description in the SDP

`ice-candidate` (gchararray) –

an ice candidate or NULL/"" to mark that no more candidates will arrive

Flags: Run Last / Action

### add-ice-candidate-full

    g_signal_emit_by_name (object, "add-ice-candidate-full", mline_index, ice-candidate, promise);

    ret = object.emit ("add-ice-candidate-full", mline_index, ice-candidate, promise)

    let ret = object.emit ("add-ice-candidate-full", mline_index, ice-candidate, promise);

Variant of the `add-ice-candidate` signal, allowing the call site to be notified using a GstPromise when the task has completed.

Parameters:

`object` (GstElement *) –

the webrtcbin

`mline_index` (guint) –

the index of the media description in the SDP

`ice-candidate` (gchararray) –

an ice candidate or NULL/"" to mark that no more candidates will arrive

`promise` (GstPromise *) –

a GstPromise to be notified when the task is complete

Flags: Run Last / Action

Since : 1.24

### add-transceiver

    g_signal_emit_by_name (object, "add-transceiver", direction, caps, &ret);

    ret = object.emit ("add-transceiver", direction, caps)

    let ret = object.emit ("add-transceiver", direction, caps);

Parameters:

`object` (GstElement *) –

the webrtcbin

`direction` (GstWebRTCRTPTransceiverDirection *) –

the direction of the new transceiver

`caps` (GstCaps *) –

the codec preferences for this transceiver

Returns (GstWebRTCRTPTransceiver *) –

the new GstWebRTCRTPTransceiver

Flags: Run Last / Action

### add-turn-server

    g_signal_emit_by_name (object, "add-turn-server", uri, &ret);

    ret = object.emit ("add-turn-server", uri)

    let ret = object.emit ("add-turn-server", uri);

Add a turn server to obtain ICE candidates from

Parameters:

`object` (GstElement *) –

the webrtcbin

`uri` (gchararray) –

The uri of the server of the form turn(s)://username:password@host:port

Returns (gboolean) –
No description available

Flags: Run Last / Action

### close

    g_signal_emit_by_name (object, "close", promise);

    ret = object.emit ("close", promise)

    let ret = object.emit ("close", promise);

Invoke the close procedure as specified in https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-close.

Parameters:

`object` (GstElement *) –

the webrtcbin

`promise` (GstPromise *) –

a GstPromise to be notified when the task is complete.

Flags: Run Last / Action

Since : 1.28

### create-answer

    g_signal_emit_by_name (object, "create-answer", options, promise);

    ret = object.emit ("create-answer", options, promise)

    let ret = object.emit ("create-answer", options, promise);

Parameters:

`object` (GstElement *) –

the webrtcbin

`options` (GstStructure *) –

create-answer options

`promise` (GstPromise *) –

a GstPromise which will contain the answer

Flags: Run Last / Action

### create-data-channel

    g_signal_emit_by_name (object, "create-data-channel", label, options, &ret);

    ret = object.emit ("create-data-channel", label, options)

    let ret = object.emit ("create-data-channel", label, options);

The options dictionary is the same format as the RTCDataChannelInit members outlined https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit and and reproduced below

ordered G_TYPE_BOOLEAN Whether the channal will send data with guaranteed ordering max-packet-lifetime G_TYPE_INT The time in milliseconds to attempt transmitting unacknowledged data. -1 for unset max-retransmits G_TYPE_INT The number of times data will be attempted to be transmitted without acknowledgement before dropping protocol G_TYPE_STRING The subprotocol used by this channel negotiated G_TYPE_BOOLEAN Whether the created data channel should not perform in-band chnanel announcement. If TRUE, then application must negotiate the channel itself and create the corresponding channel on the peer with the same id. id G_TYPE_INT Override the default identifier selection of this channel priority GST_TYPE_WEBRTC_PRIORITY_TYPE The priority to use for this channel

Parameters:

`object` (GstElement *) –

the webrtcbin

`label` (gchararray) –

the label for the data channel

`options` (GstStructure *) –

a GstStructure of options for creating the data channel

Returns (GstWebRTCDataChannel *) –

a new data channel object

Flags: Run Last / Action

### create-offer

    g_signal_emit_by_name (object, "create-offer", options, promise);

    ret = object.emit ("create-offer", options, promise)

    let ret = object.emit ("create-offer", options, promise);

Parameters:

`object` (GstElement *) –

the webrtcbin

`options` (GstStructure *) –

create-offer options

`promise` (GstPromise *) –

a GstPromise which will contain the offer

Flags: Run Last / Action

### get-stats

    g_signal_emit_by_name (object, "get-stats", pad, promise);

    ret = object.emit ("get-stats", pad, promise)

    let ret = object.emit ("get-stats", pad, promise);

The promise will contain the result of retrieving the session statistics. The structure will be named 'application/x-webrtc-stats and contain the following based on the webrtc-stats spec available from https://www.w3.org/TR/webrtc-stats/. As the webrtc-stats spec is a draft and is constantly changing these statistics may be changed to fit with the latest spec.

Each field key is a unique identifier for each RTCStats (https://www.w3.org/TR/webrtc/#rtcstats-dictionary) value (another GstStructure) in the RTCStatsReport (https://www.w3.org/TR/webrtc/#rtcstatsreport-object). Each supported field in the RTCStats subclass is outlined below.

Each statistics structure contains the following values as defined by the RTCStats dictionary (https://www.w3.org/TR/webrtc/#rtcstats-dictionary).

"timestamp" G_TYPE_DOUBLE timestamp the statistics were generated "type" GST_TYPE_WEBRTC_STATS_TYPE the type of statistics reported "id" G_TYPE_STRING unique identifier

RTCCodecStats supported fields (https://w3c.github.io/webrtc-stats/#codec-dict*)

"payload-type" G_TYPE_UINT the rtp payload number in use "clock-rate" G_TYPE_UINT the rtp clock-rate

RTCRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#streamstats-dict*)

"ssrc" G_TYPE_STRING the rtp sequence src in use "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream "kind" G_TYPE_STRING either "audio" or "video", depending on the associated transceiver (Since: 1.22)

RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)

"packets-received" G_TYPE_UINT64 number of packets received (only for local inbound) "packets-lost" G_TYPE_INT64 number of packets lost "packets-discarded" G_TYPE_UINT64 number of packets discarded "packets-repaired" G_TYPE_UINT64 number of packets repaired "jitter" G_TYPE_DOUBLE packet jitter measured in seconds

RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)

"remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound) "packets-duplicated" G_TYPE_UINT64 number of packets duplicated "fir-count" G_TYPE_UINT FIR packets sent by the receiver "pli-count" G_TYPE_UINT PLI packets sent by the receiver "nack-count" G_TYPE_UINT NACK packets sent by the receiver

RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*)

"local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds "fraction-lost" G_TYPE_DOUBLE fraction packet loss

RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*)

"packets-sent" G_TYPE_UINT64 number of packets sent (only for local outbound) "bytes-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)

RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*)

"remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats (optional since 1.22) "fir-count" G_TYPE_UINT FIR packets received by the sender "pli-count" G_TYPE_UINT PLI packets received by the sender "nack-count" G_TYPE_UINT NACK packets received by the sender

RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)

"local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats "remote-timestamp" G_TYPE_DOUBLE remote timestamp the statistics were sent by the remote

RTCPeerConnectionStats supported fields (https://w3c.github.io/webrtc-stats/#pcstats-dict*) (Since: 1.24)

"data-channels-opened" G_TYPE_UINT number of unique data channels that have entered the 'open' state "data-channels-closed" G_TYPE_UINT number of unique data channels that have left the 'open' state

RTCIceCandidateStats supported fields (https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*) (Since: 1.22)

"transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream "address" G_TYPE_STRING address of the candidate, allowing for IPv4, IPv6 and FQDNs "port" G_TYPE_UINT port number of the candidate "candidate-type" G_TYPE_STRING RTCIceCandidateType "priority" G_TYPE_UINT calculated as defined in RFC 5245 "protocol" G_TYPE_STRING Either "udp" or "tcp". Based on the "transport" defined in RFC 5245 "relay-protocol" G_TYPE_STRING protocol used by the endpoint to communicate with the TURN server. Only present for local candidates. Either "udp", "tcp" or "tls" "url" G_TYPE_STRING URL of the ICE server from which the candidate was obtained. Only present for local candidates "foundation" G_TYPE_STRING ICE foundation as defined in RFC 5245 section 15.1 (Since: 1.28) "related-address" G_TYPE_STRING ICE rel-addr as defined in RFC 5245 section 15.1. Only set for server-reflexive, peer-reflexive and relay candidates (Since: 1.28) "related-port" G_TYPE_UINT ICE rel-port as defined in RFC 5245 section 15.1. Only set for serverreflexive, peer-reflexive and relay candidates (Since: 1.28) "username-fragment" G_TYPE_STRING ICE username fragment as defined in RFC 5245 section 7.1.2.3 (Since: 1.28) "tcp-type" G_TYPE_STRING ICE candidate TCP type as defined in RTCIceTcpCandidateType (Since: 1.28)

RTCIceCandidatePairStats supported fields (https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*) (Since: 1.22)

"local-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair. "remote-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair.

Parameters:

`object` (GstElement *) –

the webrtcbin

`pad` (GstPad *) –

A GstPad to get the stats for, or NULL for all

`promise` (GstPromise *) –

a GstPromise for the result

Flags: Run Last / Action

### get-transceiver

    g_signal_emit_by_name (object, "get-transceiver", idx, &ret);

    ret = object.emit ("get-transceiver", idx)

    let ret = object.emit ("get-transceiver", idx);

Parameters:

`object` (GstElement *) –

the webrtcbin

`idx` (gint) –

The index of the transceiver

Returns (GstWebRTCRTPTransceiver *) –

the GstWebRTCRTPTransceiver, or NULL

Flags: Run Last / Action

Since : 1.16

### get-transceivers

    g_signal_emit_by_name (object, "get-transceivers", &ret);

    ret = object.emit ("get-transceivers")

    let ret = object.emit ("get-transceivers");

Parameters:

`object` (GstElement *) –

the webrtcbin

Returns (GArray *) –

a GArray of GstWebRTCRTPTransceiver

Flags: Run Last / Action

### set-local-description

    g_signal_emit_by_name (object, "set-local-description", desc, promise);

    ret = object.emit ("set-local-description", desc, promise)

    let ret = object.emit ("set-local-description", desc, promise);

Parameters:

`object` (GstElement *) –

the webrtcbin

`desc` (GstWebRTCSessionDescription *) –

a GstWebRTCSessionDescription description

`promise` (GstPromise *) –

a GstPromise to be notified when it's set

Flags: Run Last / Action

### set-remote-description

    g_signal_emit_by_name (object, "set-remote-description", desc, promise);

    ret = object.emit ("set-remote-description", desc, promise)

    let ret = object.emit ("set-remote-description", desc, promise);

Parameters:

`object` (GstElement *) –

the webrtcbin

`desc` (GstWebRTCSessionDescription *) –

a GstWebRTCSessionDescription description

`promise` (GstPromise *) –

a GstPromise to be notified when it's set

Flags: Run Last / Action

## Properties

### bundle-policy

    “bundle-policy” GstWebRTCBundlePolicy *

The policy to apply for bundling

Flags : Read / Write

Default value : none (0)

### connection-state

    “connection-state” GstWebRTCPeerConnectionState *

The overall connection state of this element

Flags : Read

Default value : new (0)

### current-local-description

    “current-local-description” GstWebRTCSessionDescription *

The local description that was successfully negotiated the last time the connection transitioned into the stable state

Flags : Read

### current-remote-description

    “current-remote-description” GstWebRTCSessionDescription *

The last remote description that was successfully negotiated the last time the connection transitioned into the stable state plus any remote candidates that have been supplied via addIceCandidate since the offer or answer was created

Flags : Read

### http-proxy

    “http-proxy” gchararray

A HTTP proxy for use with TURN/TCP of the form http://[username:password@]hostname[:port][?alpn=<alpn>]</alpn>

Flags : Read / Write

Default value : NULL

Since : 1.22

### ice-agent

    “ice-agent” GstWebRTCICE *

The WebRTC ICE agent

Flags : Read / Write / Construct Only

### ice-connection-state

    “ice-connection-state” GstWebRTCICEConnectionState *

The collective connection state of all ICETransport's

Flags : Read

Default value : new (0)

### ice-gathering-state

    “ice-gathering-state” GstWebRTCICEGatheringState *

The collective gathering state of all ICETransport's

Flags : Read

Default value : new (0)

### ice-transport-policy

    “ice-transport-policy” GstWebRTCICETransportPolicy *

The policy to apply for ICE transport

Flags : Read / Write

Default value : all (0)

### latency

    “latency” guint

Default duration to buffer in the jitterbuffers (in ms)

Flags : Read / Write

Default value : 200

Since : 1.18

### local-description

    “local-description” GstWebRTCSessionDescription *

The local SDP description in use for this connection. Favours a pending description over the current description

Flags : Read

### pending-local-description

    “pending-local-description” GstWebRTCSessionDescription *

The local description that is in the process of being negotiated plus any local candidates that have been generated by the ICE Agent since the offer or answer was created

Flags : Read

### pending-remote-description

    “pending-remote-description” GstWebRTCSessionDescription *

The remote description that is in the process of being negotiated, complete with any remote candidates that have been supplied via addIceCandidate since the offer or answer was created

Flags : Read

### remote-description

    “remote-description” GstWebRTCSessionDescription *

The remote SDP description to use for this connection. Favours a pending description over the current description

Flags : Read

### reuse-source-pads

    “reuse-source-pads” gboolean

When set to FALSE, if a transceiver becomes send-only or inactive then pre-existing source pads will receive an EOS event and no further traffic even after further renegotiation. When TRUE, pads will simply not receive any output when the negotiated transceiver state doesn't have incoming traffic. If renegotiated later, the pad will receive data again.

Flags : Read / Write

Default value : false

Since : 1.26

### sctp-transport

    “sctp-transport” GstWebRTCSCTPTransport *

The WebRTC SCTP Transport

Flags : Read

Since : 1.20

### signaling-state

    “signaling-state” GstWebRTCSignalingState *

The signaling state of this element

Flags : Read

Default value : stable (0)

### stun-server

    “stun-server” gchararray

The STUN server of the form stun://hostname:port

Flags : Read / Write

Default value : NULL

### turn-server

    “turn-server” gchararray

The TURN server of the form turn(s)://username:password@host:port. To use time-limited credentials, the form must be turn(s)://timestamp:username:password@host:port. Please note that the ':' character of the 'timestamp:username' and the 'password' encoded by base64 should be escaped to be parsed properly. This is a convenience property, use add-turn-server if you wish to use multiple TURN servers

Flags : Read / Write

Default value : NULL

## GstWebRTCBinPad

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstPad
                    ╰──GstProxyPad
                        ╰──GstGhostPad
                            ╰──GstWebRTCBinPad

### Properties

#### transceiver

    “transceiver” GstWebRTCRTPTransceiver *

Transceiver associated with this pad

Flags : Read

## GstWebRTCBinSinkPad

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstPad
                    ╰──GstProxyPad
                        ╰──GstGhostPad
                            ╰──GstWebRTCBinPad
                                ╰──GstWebRTCBinSinkPad

Since : 1.22

### Properties

#### msid

    “msid” gchararray

The MediaStream Identifier to use for this pad (MediaStreamTrack). Fallback is the RTP SDES cname value if not provided.

Flags : Read / Write

Default value : NULL

Since : 1.22

## GstWebRTCBinSrcPad

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstPad
                    ╰──GstProxyPad
                        ╰──GstGhostPad
                            ╰──GstWebRTCBinPad
                                ╰──GstWebRTCBinSrcPad

Since : 1.22

### Properties

#### msid

    “msid” gchararray

The MediaStream Identifier the remote peer used for this pad (MediaStreamTrack). Will be NULL if not advertised in the remote SDP.

Flags : Read

Default value : NULL

Since : 1.22

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